5. Playback encoded files in tested players and capture output to wav file, resolution 24bit/96kHz if possible. The problem here is that I don't know any tool for this, I asked for advice in this thread. Because of this I've only tested this on foobar2000 0.5 beta so far, as its diskwriting feature is capable of 24bit/96kHz output. So until now it's not a player output comparison but a encoder comparison as you'll see soon.

6. Aply logarithmic fadein 0 -> +150db to the whole sample with CEP to bring the volume back to original level everywhere.

7. Listening to the resulting files.

The reason for this proceeding was to "exaggerate" the differences that occur at very low volume and make them audible.

My observations: All files started to contain aditional noise compared to the original after ~1/2 of playing time. At this point started what surprised me: The mp3 sounded like the original + increasing noise till the end of the sample while MPC and Vorbis started with slight artifacts and sounded more and more awfull: ogg somehow like underwater, mpc like NUMLock's lame audiophile preset and worse.

So is mp3 superior at low volumes? Not really. I gave mp3 an advantage by using CBR. A quick test using lame ape (=VBR) showed the same awfulness as Vorbis and mpc. Besides, I noticed that with decreasing volume the bitrate of all 3 codes decreased too.

Conclusion? I'm not sure, but maybe there's something that could be improved (adaptive ath model for high resolution signals, or ...). Imagine, you have a high resolution source (DVD audio ...) and music that uses its huge dynamic range (not completely of course) - and equipment with really high SNR ... this could be an issue.

I hope I haven't just wased my time and space on this forum . Thoughts welcome.

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Let's suppose that rain washes out a picnic. Who is feeling negative? The rain? Or YOU? What's causing the negative feeling? The rain or your reaction? - Anthony De Mello